<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:courier,monaco,monospace,sans-serif;font-size:12pt">send me privately the login and pass and any other credentials I need to reproduce and a number i can call to test it<br>and i will check into it tommorow.<br><br><br><div> </div><div>Anthony Minessale II<br><br><span>FreeSWITCH <a target="_blank" href="http://www.freeswitch.org/">http://www.freeswitch.org/</a></span><br><span>ClueCon <a target="_blank" href="http://www.cluecon.com/">http://www.cluecon.com/</a></span><br><br>AIM: anthm<br>MSN:anthony_minessale@hotmail.com<br>JABBER:anthony.minessale@gmail.com<br>IRC: irc.freenode.net #freeswitch</div><div><br>FreeSWITCH Developer Conference<br>sip:888@conference.freeswitch.org<br>iax:guest@conference.freeswitch.org/888<br>googletalk:conf+888@conference.freeswitch.org<br>pstn:213-799-1400</div><div style="font-family: courier,monaco,monospace,sans-serif; font-size:
12pt;"><br><br><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;">----- Original Message ----<br>From: Ivan C Myrvold <ivan@myrvold.org><br>To: freeswitch-users@lists.freeswitch.org<br>Sent: Sunday, May 6, 2007 6:11:54 AM<br>Subject: [Freeswitch-users] Jitter problems<br><br><div>I am trying to get Freeswitch to work with my SIP provider, IP24, but <br>I always get very bad sound in one direction.<br>I made a wireshark trace, and looked at the jitter graph it produced.<br><br>I made an incoming call from my mobile phone to Freeswitch via SIP <br>provider IP24, PCMA codec:<br><br>1. <a target="_blank" href="http://www.myrvold.org/freeswitch/me2ip24.jpg">http://www.myrvold.org/freeswitch/me2ip24.jpg</a> , from freeswitch->ip24<br>2. <a target="_blank" href="http://www.myrvold.org/freeswitch/ip242me.jpg">http://www.myrvold.org/freeswitch/ip242me.jpg</a> , from ip24->freeswitch<br><br>There is a big difference in
jitter between the two graphs, and the <br>speech quality is very bad ip24->freeswitch (2).<br><br>What can I do to fix the speech quality?<br>Asterisk and OpenPBX (CallWeaver) have both excellent audio quality <br>in calls through my SIP provider IP24.<br><br><br>Here is a voip graph of the SIP call: <a target="_blank" href="http://www.myrvold.org/">http://www.myrvold.org/</a> <br>freeswitch/graph.jpg<br><br>Ivan<br><br><br><br>_______________________________________________<br>Freeswitch-users mailing list<br>Freeswitch-users@lists.freeswitch.org<br><a target="_blank" href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a target="_blank" href="http://lists.freeswitch.org/mailman/options/freeswitch-users">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a target="_blank"
href="http://www.freeswitch.org">http://www.freeswitch.org</a><br></div></div><br></div></div><br>
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